Establishing call-based audio sockets within a componentized voice server

ABSTRACT

A method of interfacing a telephone application server and a speech engine can include the step of establishing one or more audio sockets in a media converting component of the telephone application server. The audio socket can remain available for approximately a duration of a call. A work unit that requires processing by a speech engine can be detected for the call. An identifier for the audio socket and a data for the work unit can be conveyed to a selected speech engine. Work unit results from the selected speech engine can be received by the media converting component via the previously established audio socket.

BACKGROUND

1. Field of the Invention

The present invention relates to the field of telecommunications and,more particularly, to a telecommunications voice server that establishescall-based audio sockets.

2. Description of the Related Art

The Websphere Application Server (WAS) by International BusinessMachines, Corporation (IBM) of Armonk, N.Y. can be utilized by atelecommunications voice server. When so utilized, WAS can handle amultitude of telephony related tasks, a few of which can requireservices of external speech engines. The speech engines can performspeech-to-text conversions, text-to-speech conversions, and otherautomated speech related functions for the WAS.

Many speech engines, such as the IBM automatic speech recognition (ASR)engine, can use customizable dynamic link libraries (DLLs) to definedifferent audio sources. The use of customizable DLLs permits the speechengines to modularly handle a breadth of different audio sources,different audio formats, and different audio codecs. Using the DLLs, thespeech engines can act as audio socket servers, dynamically establishingports for exchanging information with external components. Further, thespeech engines can include application program interfaces (APIs) forfacilitating information exchanges. For example, the IBM ASR engineincludes an API called the Speech Manager API (SMAPI), which can be usedby the WAS to communicate with the IBM ASR. More specifically, atelephony and media (T&M) subsystem of the WAS can interface with theIBM ASR via SMAPI, where the T&M subsystem is generally responsible forperforming media conversions between the WAS and a telephony gateway,between the WAS and speech engines, and/or between the speech enginesand the telephony gateway.

In operation, a telephony call can be received that requires WASoperations. In response to call establishment, the WAS can beinitialized. Initialization includes activating the T&M subsystem todetect audio utterances occurring within the established call. When anutterance is detected, the T&M subsystem can briefly cache the utteranceas the WAS determines appropriate actions to perform. One possibleaction involves speech-to-text converting the utterance. To perform thisconversion, the WAS assigns a speech engine to handle the utterance. Thespeech engine dynamically establishes an audio socket. An identifier forthe audio socket is conveyed through the WAS to the T&M subsystem. Uponreceiving the identifier, the T&M subsystem conveys the utterance to theselected speech engine via the established audio socket. Once theutterance has been processed by the speech engine, the connectionbetween the T&M subsystem is terminated and the audio socket is closedand/or reallocated for other processing tasks.

It should be appreciated that the WAS, like most high volume servers,performs turn based speech engine allocations as opposed to call basedallocations. Turn based allocation techniques dynamically assigndiscrete work units or turns to speech engines as needed. Call basedallocation techniques provide a 1-1 speech engine to telephone callmapping. As speech engines are typically costly and consume extensivecomputing resources, cost effective telephony solutions do not generallyperform call-based allocation, but rather perform turn-based allocationof speech engines, thereby maximizing the usage of expensive speechengine components.

The aforementioned approach for utilizing speech engines, however, canbe problematic. One such problem is that numerous turns for processingdifferent utterances are commonly performed during each telephone call.For each turn, the T&M subsystem conveys audio signals to a particularspeech engine via a specified audio socket. Accordingly, throughout thecall, the T&M subsystem handles continuously changing audio ports thatare dynamically allocated by the various speech engines. Moreover, eachtime a speech engine allocates an audio socket, the host/port/protocolfor the audio socket established by the speech engine must be conveyedto the T&M subsystem before audio signals can be conveyed between theT&M subsystem and the speech engine.

Conveying the audio socket information from the speech engine to the T&Msubsystem can result in processing delays. These delays can bepronounced when the voice server through which the socket information isconveyed has a componentized and functionally isolated architecture, asdoes the WAS. Appreciably, such an architecture does not constantlymaintain a call-based control path between the T&M subsystem and thespeech engine. A skilled artesian can recognize that this approach issubject to numerous bottlenecks which can be problematic when the voiceserver, T&M subsystem, and/or the speech engines are placed undersignificant loads. Consequently, it would be highly advantageous toutilize a different approach that reduces latencies resulting from thesebottlenecks.

SUMMARY OF THE INVENTION

The present invention includes a method, a system, and an apparatus forestablishing call-based audio sockets within a componentized voiceserver in accordance with the inventive arrangements disclosed herein.More specifically, a media converting component of a voice server, suchas the telephony and media (T&M) subsystem of a Websphere applicationserver (WAS), can function as an audio socket server. When a call isinitialized, one or more audio sockets can be established by the mediaconverting component for approximately the duration of the call. Anidentifier for the established sockets can be conveyed to othercomponents of the voice server along with additional call specificinformation. When the voice server requires services of a remote speechserver, the identifier for the previously established audio socket canbe conveyed to the selected speech engine along with other datanecessary for performing the desired service. The speech engine canestablish a communication link with the media converting component viathe identified socket. Data can be conveyed between the media convertingcomponent and the speech engine as appropriate. Once the speech enginehas completed its assigned tasks, the connection with the socket can beterminated, yet the audio socket can remain open for othercommunications with other speech engines for the duration of the call.

One aspect of the present invention includes a method of interfacing atelephone application server and a speech engine. The method can includethe step of establishing one or more audio sockets in a media convertingcomponent of the telephone application server. The audio socket canremain available for approximately a duration of a call. A work unitthat requires processing by a speech engine can be detected for thecall. An identifier for the audio socket and a data for the work unitcan be conveyed to a selected speech engine. Work unit results from theselected speech engine can be received by the media converting componentvia the previously established audio socket.

It should be notated that the invention can be implemented as a programfor controlling a computer to implement the functions described herein,or a program for enabling a computer to perform the processcorresponding to the steps disclosed herein. This program may beprovided by storing the program in a magnetic disk, an optical disk, asemiconductor memory, any other recording medium, or distributed via anetwork.

Another aspect of the present invention includes a system for providingspeech services that includes a telephone application server, at leastone speech engine, and a telephone gateway. The telephone applicationserver can have a componentized architecture of different modularcomponents. One of these modular components can include a mediaconverting component that can function as an audio socket server. Thespeech engine can communicate to the telephone gateway using the mediaconverting component as an intermediary. The speech engine can connectto the telephone gateway via sockets established by the speech engine,where the established sockets can be associated with selected telephonecalls. The sockets can remain available for approximately the durationof the associated telephone call.

BRIEF DESCRIPTION OF THE DRAWINGS

There are shown in the drawings, embodiments that are presentlypreferred; it being understood, however, that the invention is notlimited to the precise arrangements and instrumentalities shown.

FIG. 1 is a schematic diagram illustrating a system that includes avoice server that provides speech services in accordance with theinventive arrangements disclosed herein.

FIG. 2 is a method for implementing telecommunication speech services inaccordance with the inventive arrangements disclosed herein.

DETAILED DESCRIPTION OF THE INVENTION

FIG. 1 is a schematic diagram illustrating a system 100 that providesspeech services in accordance with the inventive arrangements disclosedherein. The system 100 can include a telephone gateway 115, acomponentized voice server 150, and a multitude of speech engines 130.The telephone gateway 115 can include hardware or software thattranslates protocols and/or routes calls between a telephone network110, such as a Public Switched Telephone Network (PSTN), and the voiceserver components 150. For example, the telephone gateway 115 caninclude a Cisco 2600 series router from Cisco Systems, Inc. of San Jose,Calif., a Cisco, a Cisco 5300 series gateway, a Digital Trunk eXtendedAdapter (DTXA), a Intel (R) Dialogic (R) Adaptor from Intel Corporationof Santa Clara, Calif., and the like.

The speech engines 130 can include one or more automatic speechrecognition engines 134, one or more text to speech engines 132, andother speech related engines and/or services. Particular ones of thespeech engines 130 can include one or more application programinterfaces (APIs) 136 for facilitating communications between the speechengine 130 and external components. For example, in one embodiment, theASR engine 134 can include an IBM ASR engine with an API such as aSpeech Manager API (SMAPI).

The voice server 150 can have a componentized and isolated architecturethat can include voice server components 155 and a media convertercomponent 125. The voice server components 155 can include a telephoneserver, a dialogue server, a speech server, one or more web servers, andother such components. Selective ones of the voice server components 155can be implemented as Virtual Machines, such as virtual machinesadhering to the JAVA 2 Enterprise Edition (J2EE) specification. In oneembodiment, the voice sever 150 can be implemented using the WebsphereApplication Server (WAS), where the WAS is a particular implementationof J2EE. In another embodiment, a call descriptor object (CDO) can beused to convey call data between the voice server components 155. Forexample, the CDO can specify the gateway identifiers, audio socketidentifiers, telephone identification data, and/or the like.

The media converter 125 can perform media conversions between thetelephone gateway 115 and the speech engines 130, between the voiceserver components 155 and the telephone gateway 115, and between thevoice server components 155 and the speech engine 130. In oneembodiment, the media converter 125 can be a centralized interfacingsubsystem of the voice server 150 for inputting and outputting data toand from the voice server 155. For example, the media converter 125 caninclude a T&M subsystem,.such as the T&M subsystem of a WAS.

The media converter 125 can establish a plurality of media ports 122,thereby functioning as an audio socket server. Each of the media ports122 can be used to establish a communication link with a speech engine130. Moreover, each of the media ports 122 can be associated with atelephone call. When a call is initialized, one or more media ports 122can be established. The media ports 122 can remain active and availablefor approximately the duration of the call. Multiple different speechengines 130 that process different turns for a call can use the samemedia port 122 to relay information to and from the media converter 125.

The system 100 can also include a resource connector 120. The resourceconnector 120 can be a communication intermediary between the telephonegateway 115 and the voice server components 155 and/or media converter125. The resource connector 120 can manage resource allocations forcalls.

In operation, a user can initiate a telephone call. The call can beconveyed through the telephone network 110 and can be received by thetelephone gateway 115. The telephone gateway 115, having performed anyappropriate data conversions, can convey call information to theresource connector 120. For example, call information can be conveyedusing a session initiation protocol (SIP). In particular embodiments,the telephone gateway 115 can also convert circuit-switched data topacket-switched data for processing by the media converter 125 and thevoice server 150. In other embodiments, the resource connector 120 canconvert circuit-switched data to packet-switched data as appropriate.The resource connector 120 can generate a CDO that contains call relatedinformation, including the port(s) that telephone gateway 115 hasassigned to the call. In one embodiment, the CDO can be a Java objectand the assigned port(s) can include Reliable Data Protocol (RDP)port(s).

Once generated, the CDO can be sent to the media converter 125, whichcan establish one or more media ports 122 that can be used for the call.Identifiers, which can be Uniform Resource Identifiers (URI), associatedwith the established media ports 122 can be added to the CDO. The CDOcan then be conveyed to voice server components 155 as appropriate forconducting the operations of the voice server 150. The voice servercomponents 155 can determine that one or more work units need processingby a speech engine 130. For each work unit, an appropriate speech engine130 can be selected. The CDO can be conveyed to the speech engine 130.

The speech engine 130 can connect to the media converter 125 via themedia port 122 identified within the CDO. The speech engine 130 can thenbe conveyed appropriate input signals over the established connection.The speech engine 130 can process the work unit and convey work unitresults back to the media converter 125 over the established connection.After the speech engine 130 has handled the work unit, the connectionwith the media converter 125 can be terminated so that other speechengines 130 can utilize the open media port 122 for conveying datarelating to other work units.

Accordingly, the present invention can establish a call-basedcommunication path between the media converter 125 and the speechengines 130. That is, the media ports established for the call by themedia converter 125 remain available to the various speech engines 130throughout the duration of the call. Consequently, the speech engines130 need not dynamically establish new communication sockets as workunits are processed, nor do the speech engines 130 need to communicatesocket establishment port information to the media converter 125. Thus,the numerous bottlenecks and latencies present when speech enginesdynamically establish sockets on a work unit basis are not presentwithin system 100.

FIG. 2 is a method 200 for implementing telecommunication speechservices in accordance with the inventive arrangements disclosed herein.The method 200 can be performed in the context of a voice server havinga componentized and functionally isolated architecture. One of thesecomponents can be a telecommunications and media interface componentthat functions as a media converter. The method can begin in step 205,where a telephone gateway can receive an incoming call. In step 210, acomponentized voice server can be initialized to handle the call. Instep 215, a media converter, which is a component of the voice server,can establish one or more audio ports for the call. In step 220, calland media converter information can be conveyed to other components ofthe voice server as needed. For example, a Uniform Resource Identifier(URI) that identifies the audio ports established by the media convertercan be included within a software object that is conveyed between voiceserver components. In such an example, additional call information, suchas telephony gateway ports, calling and called telephone numbers, callidentifying data, and the like can be included within the conveyedsoftware object.

In step 225, the voice server can determine that a.turn needs processedfor the call. The turn is a designated work unit that is typically basedupon a speech utterance text segment that is to be converted into aspeech utterance. In step 230, the voice server can determine a speechengine for handling the turn. In step 235, the voice server can conveycall and media converter information to the selected speech engine. Instep 240, the speech engine can establish a connection via thepreviously established audio port. In step 245, the speech engine canconvey turn results to one of the previously established ports of themedia converter. In step 250, a determination can be made as to whetherthe call is complete. If so, the method can loop back to step 225, wherethe voice server can determine that another turn needs processing. Ifthe call is complete, the method can proceed to step 255, where themedia. converter can close the established audio ports. The telephonegateway can also close the ports that it established for the call. Thatis, the audio and gateway ports can be closed responsive to callcompletion. Thus, by establishing call-based sockets when a call isestablished, latencies typical of speech engines that function as socketservers can be eliminated or at least considerably reduced.

The present invention can be realized in hardware, software, or acombination of hardware and software. The present invention can berealized in a centralized fashion in one computer system or in adistributed fashion where different elements are spread across severalinterconnected computer systems. Any kind of computer system or otherapparatus adapted for carrying out the methods described herein issuited. A typical combination of hardware and software can be ageneral-purpose computer system with a computer program that, when beingloaded and executed, controls the computer system such that it carriesout the methods described herein.

The present invention also can be embedded in a computer programproduct, which comprises all the features enabling the implementation ofthe methods described herein, and which when loaded in a computer systemis able to carry out these methods. Computer program in the presentcontext means any expression, in any language, code or notation, of aset of instructions intended to cause a system having an informationprocessing capability to perform a particular function either directlyor after either or both of the following: a) conversion to anotherlanguage, code or notation; b) reproduction in a different materialform.

This invention can be embodied in other forms without departing from thespirit or essential attributes thereof. Accordingly, reference should bemade to the following claims, rather than to the foregoingspecification, as indicating the scope of the invention.

1. A method of interfacing a voice server and a speech engine comprisingthe steps of: establishing at least one audio socket in a mediaconverting component of a voice server, said audio socket remainingavailable for approximately a duration of a call; detecting a work unitof said call that requires processing by a speech engine; conveying anidentifier for the audio socket and data relating to a work unit to aselected speech engine; and receiving work unit results from theselected speech engine via the audio socket.
 2. The method of claim 1,wherein the work unit is a speech utterance of a party participating inthe call, and wherein the speech engine is an automatic speechrecognition engine.
 3. The method of claim 1, wherein the work unit is atext segment generated by the telephone application server, and whereinthe speech engine is a text-to-speech engine.
 4. The method of claim 1,further comprising the steps of: after the work unit data is processed,the speech engine initializing a communication connection with the audiosocket; and responsive to said initialization, conveying pending audiosignals from the media converting component to the audio socket.
 5. Themethod of claim 1, further comprising the steps of: detecting adifferent work unit of said call that requires processing by a speechengine; conveying said identifier for the audio socket and conveyingdata relating to the different work unit to a different speech engine;and receiving work unit results from said different speech engine to theaudio socket.
 6. The method of claim 5, wherein said speech engine is anautomatic speech recognition engine and wherein said different speechengine is a text-to-speech engine.
 7. The method of claim 1, whereinsaid at least one audio socket is a plurality of audio sockets.
 8. Themethod of claim 7, wherein said plurality of audio sockets comprises atleast one input audio socket and at least one output audio socket. 9.The method of claim 1, wherein said telephone application server has acomponentized architecture of different modular components.
 10. Themethod of claim 9, wherein each component within said architecture isconfigured to handle tasks in a functionally isolated fashion from othercomponents of said architecture.
 11. A machine-readable storage havingstored thereon, a computer program having a plurality of code sections,said code sections executable by a machine for causing the machine toperform the steps of: establishing at least one audio socket in a mediaconverting component of a voice server, said audio socket remainingavailable for approximately a duration of a call; detecting a work unitof said call that requires processing by a speech engine; conveying anidentifier for the audio socket and a work unit data to a selectedspeech engine; and receiving work unit results from the selected speechengine via the audio socket.
 12. The machine-readable storage of claim11, further comprising the steps of: after the work unit data isprocessed, the speech engine initializing a communication connectionwith the audio socket; and responsive to said initialization, conveyingpending audio signals from the media converting component to the audiosocket.
 13. The machine-readable storage of claim 11, further comprisingthe steps of: detecting a different work unit of said call that requiresprocessing by a speech engine; conveying said identifier for the audiosocket and conveying data relating to the different work unit to adifferent speech engine; and receiving work unit results from saiddifferent speech engine to the audio socket.
 14. The machine-readablestorage of claim 13, wherein said speech engine is an automatic speechrecognition engine and wherein said different speech engine is atext-to-speech engine.
 15. The machine-readable storage of claim 11,wherein said at least one audio socket is a plurality of audio sockets.16. The machine-readable storage of claim 15, wherein said plurality ofaudio sockets comprises at least one input audio socket and at least oneoutput audio socket.
 17. The machine-readable storage of claim 11,wherein said telephone application server has a componentizedarchitecture of different modular components, wherein each componentwithin said architecture is configured to handle tasks in a functionallyisolated fashion from other components of said architecture.
 18. Asystem for providing speech services comprising: a telephone applicationserver having a componentized architecture of different modularcomponents, wherein one of said components is a media convertingcomponent configured as an audio socket server; at least one speechengine, each configured to perform at least one programmatic action forsaid telephone application server; and a telephone gateway thatcommunicatively links said telephone application server to a telephonenetwork.
 19. The system of claim 18, wherein said speech enginecommunicates to said telephone gateway using said media convertingcomponent as an intermediary, and wherein said speech engine connects tosaid telephone gateway via sockets established by the media convertingcomponent, said established sockets being associated with selectedtelephone calls, and wherein said sockets remain available forapproximately the duration of the associated telephone call.
 20. Thesystem of claim 19, wherein different speech engines convey work unitresults via a single one of said sockets during a telephone call forwhich the socket was established.